WO2001005076A1 - Procede de transfert d'information dans un systeme de telecommunications - Google Patents

Procede de transfert d'information dans un systeme de telecommunications Download PDF

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Publication number
WO2001005076A1
WO2001005076A1 PCT/SE2000/000898 SE0000898W WO0105076A1 WO 2001005076 A1 WO2001005076 A1 WO 2001005076A1 SE 0000898 W SE0000898 W SE 0000898W WO 0105076 A1 WO0105076 A1 WO 0105076A1
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WO
WIPO (PCT)
Prior art keywords
estimation
transceiver
time
receiving side
sampling frequency
Prior art date
Application number
PCT/SE2000/000898
Other languages
English (en)
Inventor
Anders Nohlgren
Jim Sundquist
Original Assignee
Telefonaktiebolaget Lm Ericsson (Publ)
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Telefonaktiebolaget Lm Ericsson (Publ) filed Critical Telefonaktiebolaget Lm Ericsson (Publ)
Priority to EP00928087A priority Critical patent/EP1198910B1/fr
Priority to DE60039763T priority patent/DE60039763D1/de
Priority to AU46374/00A priority patent/AU4637400A/en
Priority to JP2001509191A priority patent/JP4553537B2/ja
Publication of WO2001005076A1 publication Critical patent/WO2001005076A1/fr

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L12/00—Data switching networks
    • H04L12/64—Hybrid switching systems
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04J—MULTIPLEX COMMUNICATION
    • H04J3/00—Time-division multiplex systems
    • H04J3/02—Details
    • H04J3/06—Synchronising arrangements
    • H04J3/062—Synchronisation of signals having the same nominal but fluctuating bit rates, e.g. using buffers
    • H04J3/0632—Synchronisation of packets and cells, e.g. transmission of voice via a packet network, circuit emulation service [CES]

Definitions

  • the invention is concerned with a method for sending information between at least two transceivers in a telecommunication system.
  • the method of the invention is suitable for transporting messages over packet based networks.
  • IP-telephony Most packed based systems of today are based on the Internet protocol (IP), and its sub protocols, the Transmission Control Protocol (TCP) and the User Datagram Protocol (UDP). TCP guarantees reliable transmission of data and allows some sort of flow control.
  • TCP Transmission Control Protocol
  • UDP User Datagram Protocol
  • IP Internet Protocol
  • TCP Transmission Control Protocol
  • UDP User Datagram Protocol
  • FTP File Transfer Protocol
  • UDP does not provide any guarantees about the connection, but is used when a guaranteed connection requires too much control signaling.
  • Real-time applications as for example IP-telephony, use UDP. For these applications, retransmission of lost packets makes no sense since resent packets will be too late to be used in the synthesis at the receiving side anyhow.
  • IP-telephony uses the Real Time Protocol (RTP) together with IP/UDP protocols.
  • RTP Real Time Protocol
  • the RTP header contains information about sequence number, the packet's time etc.
  • RTP is e.g. used to synchronize audio and video streams.
  • Another essential part of the transmission of real-time streams is the Real Time Control Protocol (RTCP). It is used for the control of RTP.
  • RTCP conveys information about the session participants, and periodically distributes control packets containing quality information to all session participants.
  • the playout buffer is a buffer where the speech samples are stored before they are played out by the D/A converter. If there is underrun, the playout buffer will get into starvation, i.e. there will no longer be any samples to play on the output. Overrun occurs when the playout buffer is filled with samples. Consequently, samples will be lost.
  • the reason for these problems is the lack of synchronization between the sampling rates (i.e. sampling frequencies) at the sending and receiving side.
  • the messages are sent in form of digital signals from the sender of a transceiver to the receiver of another transceiver.
  • the signals transmitted from the sender have a first sampling frequency.
  • the receiver buffers these signal in a playout buffer with this sampling frequency but plays them out with a second sampling frequency.
  • the first frequency with which the signals are buffered in the playout buffer
  • the second frequency which is the playout frequency
  • the first frequency is lower then the second one, the play-out buffer might come into a situation without samples, i.e. underrun.
  • the sampling frequency of all terminals connected to the network are controlled by an accurate timing reference provided by the system.
  • PLLs Phase locked loops
  • the sampling frequency can be controlled. If the buffer is growing it plays out faster, where after the buffer return to its default value. The sampling rate is all the time corrected depending on the size of the buffer.
  • time stretching could be used to give a stimulus the same duration at the receiving side as the duration the same stimuli had on the sending side. How much to stretch the signals depends on the difference in sampling frequency between the sending side and the receiving side(s). Time stretching means that a stimulus of N samples is replaced with one with M samples. By doing this time stretching in an appropriate way, overrun or underrun will never occur.
  • the object of the invention is therefore a method and arrangement that provides for faster estimation of the clock skew to avoid delays and/or interrupts in the transmission from sender to receiver.
  • information data is sent between at least two transceivers in a telecommunication system.
  • the information data is transmitted from the sender of a transceiver to the receiver of one or more other transceivers in form of digital signals having a given sampling frequency.
  • the signals are played out by the receiver in a controlled way.
  • the method is mainly characterized by estimation of the sender's sampling rate at the sending side of a transceiver, transmitting the estimation to the receiving side of an another transceiver, and controlling the playout of the information data at the receiving side by means of the sampling rate estimated at the sending side to avoid delays and/or interrupts in the presentation.
  • the invention is especially suitable in connection with packet based networks wherein the information data is sent between the transceivers in the telecommunication system in form of packet data frames, such as audio frames.
  • the transceivers comprise both a sender and a receiver, but the invention, of course, also covers such cases, wherein the information data is sent from a transceiver which only works as a sender and wherein the information data is received by a transceiver which only works as a receiver.
  • the communication is usually an interactive communication, but the invention can also be applied for example in situations, in one way communications and in communications with several transceivers.
  • the method when a two-way communication is performed between at least two transceivers, the method is performed so that an estimation of the sender's sampling frequency is performed at the sending side of a first transceiver, said estimation is transmitted to the receiving side of a second transceiver, the play-out of the received data is controlled at said receiving side of said second transceiver by means of the sampling rate estimated at said sending side of said first transceiver, an estimation of the sampling frequency of the sending side of said second transceiver is performed at said sending side of said second transceiver, the estimation of the sampling frequency of said sending side of said second transceiver is transmitted to the receiving side of said first transceiver, and the play-out of the received data is controlled at said receiving side of said first transceiver by means of the sampling rate estimated at said sending side of said second transceiver.
  • the communication between at least two transceivers can be performed so that an estimation of the sender's sampling frequency is performed at the sending side of a first transceiver, the estimation is transmitted to the receiving side of a second transceiver, the play-out of the received data is controlled at said receiving side of said second transceiver by means of the sampling rate estimated at said sending side of said first transceiver, the estimation of the sampling rate estimated at said sending side of the first transceiver is used in the transmitting of messages from the second transceiver to the first transceiver in the communication between said transceivers.
  • the controlling of the play-out of received data at said receiving side by means of the sampling rate estimated at said sending side is carried out by estimation of the receiver's sampling frequency at said receiving side and performing a compensation of the difference in said estimated sampling frequencies at said sending and receiving sides by a sample rate conversion method.
  • Said conversion method can be a method known in the art, e.g. a method, wherein the amount of samples in the packet frames are changed.
  • the method can for example be the one referred to earlier on page 2, i.e. the one presented in "Applications of Digital Signal Processing to Audio and Acoustics" (p.291) by Mark Kahrs and Karlheinz Brandenburg.
  • the controlling of the play-out of received data at said receiving side by means of the sampling rate estimated at said sending side is carried out by synchronizing the receiver's sampling rate at said receiving side to the sender's sampling rate.
  • This method usually requires a stable reference frequency.
  • the synchronization can be carried out by some known method in the art, for example by means of the earlier mentioned PPL-method used for cellular systems.
  • Said transmitting of the estimation can be done during a call-set up, which is the more preferable alternative or alternatively, during the session.
  • Said estimation is incorporated in regular reports, in which case the estimation can be incorporated in the reports sent with the RTCP protocol and/or transmitted as separate reports in own packets.
  • each terminal can continuously estimate its own sampling frequency, ; .
  • this estimation is transmitted to the other terminal/terminals.
  • the correct sampling frequency is now available at the receiving side already at call setup and it can immediately be used to control the playout buffer, i.e. there is no initial delay until the estimate is available.
  • the estimation of the sampling rate is carried out in form of a calculation based on the time measured between two events and the number of samples that has been sampled between them.
  • These events are advantageously two time synchronization events.
  • a time synchronization event can be such an event, wherein a host clock in the transceiver is e.g. synchronized to an external clock.
  • the time can also be measured between two frame delivers of packet data, e.g. audio frames.
  • a ticking Central Processing Unit (CPU) clock can be used to measure the time value between two events by calculating the number of ticks between the events.
  • the nominal number of ticks per second is given through a system constant, depending on the nominal crystal frequency, which is the frequency set on the computer processor.
  • the true number of ticks per second is estimated by means of a long term stable time reference and the computer clock.
  • the long term stable time reference can be a synchronized host clock.
  • the true number of ticks per second can be estimated as a function of the time difference between two computer clock values and the time difference between two reference time values. In one embodiment, the number of ticks per second is calculated as a moving average of the last few estimations.
  • the frequency estimation is advantageously carried out directly by means of the time difference between the time values at two synchronization events and the time difference between two reference time values at the same events.
  • the estimation can also be carried out by means of a moving average of the last few estimations.
  • the estimation process of the invention is preferably performed continuously so that the best possible estimate would be available all the time.
  • the arrangement of the invention comprising at least two transceivers in a telecommunication system is mainly characterized by means for estimation of the sender's sampling frequency at the sending side of a transceiver, means for transmitting the estimation to the receiving side of another transceiver, and means for controlling the play-out of the received data at the receiving side by means of the sampling rate estimated at said sending side to avoid delays and/or interrupts in the presentation (the playout at the receiving side).
  • said means for controlling the play-out of the received data at said receiving side comprises means for estimation of the receiver's sampling frequency and means for performing a compensation of the difference in said estimated sampling frequencies at the sending and receiving sides by means of a sample rate conversion method.
  • the means for estimation of the sampling rate at the sending side preferably comprises a Calculation Unit (CCU) for calculation the CPU ticks per seconds, and a Sampling Frequency Estimation Unit (SEU).
  • the means for transmitting the estimation to the receiving side comprises a Sampling Frequency Distribution Unit (SDU), which is the interface between the transfer protocols and said estimation units (CCU, SEU).
  • SDU Sampling Frequency Distribution Unit
  • the invention can be used to continuously estimate the sampling frequency in a terminal and transmitting this information to the receiving terminals, so that better buffer management algorithms can be deployed. Consequently, better audio quality is achieved.
  • the estimation of true number of CPU ticks per second in the host can be distributed to other applications. Those then have access to a more accurate timing information provided through the estimated CPU ticks per second.
  • the invention is especially useful in real-time applications with a periodicity, typically audio and video. These applications take a speech frame or a picture with given intervals. The delay in sending has to be kept low since they are interactive applications.
  • Figure 1 is a flow scheme of some preferred embodiments of the method of the invention
  • FIG. 2 is a schematic view of the arrangement of the invention DETAILED DESCRIPTION
  • the sample clock, f is a clock, which is generated on the soundboard, if sound is sent, and it is separated from the computers own clock.
  • the sound board is a signal receiving entity that takes samples on given time intervals.
  • the sample clock works on discrete increasing integer time values and each step corresponds to a sample interval, T s.
  • the sample interval is the inverse of the sampling frequency, F s .
  • the central processing unit (CPU) clock, f is the computer's own clock. It works on discrete, increasing integer values. One step is often called “a tick” and in this application the CPU clock is said to be a “ticking" clock.
  • An application can use the CPU clock to measure the time between two events. By reading the value of f at two different times it can calculate the number of ticks between them. The actual time between these two events can then be calculated if the number of ticks per second is known. In a computer, this is provided through a system constant, T c , which is calculated from the nominal crystal frequency in the processor. However, the true number of ticks per second, T c , is not available as the true number depends on e.g. the manufacturing and the conditions in which the processor is used.
  • the host clock gives the time in year, month, day, hour, minutes, seconds and microseconds. It is usually controlled by the CPU clock, but it can be synchronized to an external clock.
  • the synchronization can be done through the Network Time protocol (NTP).
  • NTP is used in a computer network to allow the computers in the network to synchronize their clocks to Universal time Coordinated time (UTC time).
  • UTC time the same as Greenwich Mean Time (GMT)
  • GTT Greenwich Mean Time
  • GMT Greenwich Mean Time
  • GMT Greenwich Mean Time
  • GPS Global Positioning System
  • a GPS receiver can be installed and utilized in a workstation to synchronize f to the stable time scale provided in GPS.
  • the optimal time to perform the measurements is directly after the host clock has been adjusted to the external time reference.
  • the "true” clock, t is the time provided by NTP or GPS.
  • Figure 1 is a flow scheme of some preferred embodiments a,b, and c of the invention.
  • Step 1 is common for all of the three illustrated embodiments of the invention.
  • the sampling frequency can be estimated in different ways, for example depending on if there is a possibility to poll the buffer status to find out how many samples there are available to be brought up to the application.
  • the events when the buffer deliver packet frames to the application, are used in the estimation of the sampling frequency.
  • audio frames When for example audio frames are delivered to the application, they usually have a fixed block size, for instance the number of samples needed as input to a speech coding unit.
  • An accurate and fast way in accordance with the invention to estimate the sample frequency at the sender side, e.g. the sampling frequency of the soundboard, is to make use of three unsynchronized series of events.
  • the first event is the above mentioned CPU tick counter that is continuously increasing with time.
  • the second event is the updating of the host clock by the external time reference indicated in step 2 of figure 1.
  • the third series of events is the receiving of the data packets, for example when the soundboard delivers audio frames to the application.
  • the number of CPU ticks since the last data packet delivery is estimated.
  • the best possible estimate of the number of CPU ticks per second, f c is achieved by doing the estimation at the time instances when the host clock is synchronized to the external time reference as indicated in step 2 of figure 1.
  • the estimation of the number of CPU ticks per second, f c can also be estimated during a very long period as a background job in a terminal. E.g. the counting of the number of CPU ticks per second can begin already when the terminal is booted up (not illustrated).
  • T c can be estimated as a function of to , which is the time of the host clock when the estimation process started, (step 1 of figure 1), for example the time the computer is turned on, t n h , which is the time of the host clock i 1 when it was synchronized to an external clock, as indicated in step 2 of figure 1 (step 2 is common for two of the illustrated embodiments of the invention), for example the n:th NTP or GPS upgrade at time point n, to , which is the time of the CPU clock when the estimation process started, as indicated in step 1 of figure 1 , and t n c , which is the time of the CPU, when the host clock was synchronized to an external clock, as indicated in step 2 of figure 1 as above,
  • Different algorithms can be used to calculate f c ,as indicated in step 3 of figure 1.
  • the algorithm can generally be expressed as
  • f m , f n , ⁇ m , ? n , fo and io are defined as above.
  • the subscript defines the time point and the superscript tells refers to the clock.
  • f c can be calculated as a moving average of the last few estimations according to the equation
  • k(n) is the weight of the n:th estimation (Different measurements are advantageously weighted differently depending on when the measurement was carried out. Usually the latest measurements are weighted more. The sum of the weights shall be 1), t c - t c
  • time point n-1 is the time point at the event preceding an event at time point n and thus t n c -t culinary c _, is the difference in computer clock values at these time points and t n ' -t n '_ ⁇ is the time difference in the synchronized host clock values (true time values) at the same time points.
  • the estimate of f c will not have the same accuracy.
  • the estimation process is executed continuously when the computer is turned on. By the time the call is initiated an accurate estimate of f c will be available.
  • step 5a the number of samples in a packet data is noted.
  • the number is known, as the packets are of a fixed length. If the packet size varies, the number can be figured out by reading data of how many samples each packet contains and by taking these values into consideration, which gives one more variable.
  • the sampling frequency is achieved as indicated in step 6a by means of e.g. the following function.
  • ⁇ c n is the value of the CPU clock at the delivery of the n:th packet and correspondingly ⁇ c m for the delivery of the m:th packet.
  • the estimated sampling frequency is transmitted in step 7, for example by using RTCP protocol, to this receiving side of another transceiver, whereby the packet to be transmitted to the receiver might be of the Application- defined RTCP (APP) packet type.
  • APP Application- defined RTCP
  • the packet type there is a field for application-dependent data which can contain the estimated sampling frequency, but in this case, the reports must be extended with a profile- specific extensions according to Schulzrinne, H., et AL, "RTP: A transport Protocol for Real-Time Applications", RFC 1889, IETF, January 1996.
  • step 8 the own sampling frequency is estimated at the receiver with the same methods and a compensation of the difference in said estimated sampling frequencies at said sending and receiving sides is carried out by a sample rate conversion method.
  • Said conversion method can be a method known in the art, e.g. a method, wherein the amount of samples in the packet frames are changed.
  • the method can for example be the one referred to earlier on page 2, i.e. the one presented in "Applications of Digital Signal Processing to Audio and Aucoustics" (p.291) by Mark Kahrs and Karlheinz Brandenburg.
  • the estimation can be done without making use of the time synchronization events.
  • the sampling frequency is estimated by means of the time between two different frame events as polling of the buffer status is not carried out or cannot be done depending on the construction of the operative system and the soundboard. However, if the time between host synchronization events is larger than the measurement time, the inaccuracy in the estimate will be as large as the unsynchronized host clock.
  • step 2c delivering of data, e.g. audio frames to the application, is identified.
  • the time between two frame events is then measured in step 3c, where after the sampling frequency, F s , can be estimated by calculation as indicated in step 4c, for example from the equation
  • t n h -t m h is the difference between the arriving times (host clock times) of two data frames at time point n resp.m
  • N is the amount of samples in a frame
  • n is the frame number at time point n
  • m is the frame number at time point m
  • step 2 the most accurate time for the host clock, f 1 will be available. Since f is synchronized to t, ( can be used in the equation instead of f 1 .
  • the status of the buffer is polled in step 4b and the sampling frequency can be calculated based on the time between the synchronization events and the number of samples that has been sampled between them.
  • f n is the time of the sample clock at time point n
  • fo is the time of the sample clock at time point 0, for example when the estimation started
  • f n is the time of the host clock at a synchronization event at time point n (the "true” clock)
  • fo is the time of the host clock at a synchronization event at time point 0 (the "true” clock)
  • F s can be calculated as a moving average of the last few estimations according to the equation
  • time values are defined as above, the superscript s meaning the sample clock, the superscript t meaning the true clock, and the subscripts indicating different time points explained earlier and k(n) is the weight of the n:th estimation. To get the best estimation, the calculation should be done at a host clock synchronization.
  • the clock skew between the true clock and the host clock can also be calculated. By calculating the frequency difference between these clocks, this information can be used to adjust the read host time based on the time since the last host clock correction. If the frequency difference between the true clock and the host clock is calculated by means of the adjusted host time, the sampling frequency can be calculated. With the information about the clock skew, the time, measured with the host clock between frame n and m, can be compensated to yield true time. As in the case of estimating f c , this process needs to be performed continuously to get the best estimation.
  • N is the amount of samples k is the correction factor between real-time and host clock time and the time difference is defined as above.
  • FIG. 2 presents an arrangement of the invention to carry out the claimed method.
  • the arrangement comprises a CPU ticks Calculating Unit (CCU).
  • the calculation unit has the reference number 1 in figure 2.
  • the calculation unit 1 uses a stable time reference, supplied through an external source , e.g. NTP or GPS, to calculate an estimate of the number of CPU ticks per second, T c .
  • the inputs to the Calculation Unit is a long term stable time reference and f, which is the value of the CPU clock as described above.
  • the time reference can for example be the value of a synchronized host clock, f, as described above. If NTP or GPS is used to synchronize f 1 to , the time base provided by the host clock will have a good long term accuracy.
  • the output from the CCU is f c
  • the estimated T c is forwarded to a Sampling Frequency Estimation Unit
  • the Sampling Frequency Estimation Unit has the reference number 2 in figure 2.
  • the estimate of the sampling frequency is forwarded to the Sampling Frequency Distribution unit (SDU) indicated with reference number 3 in figure 2.
  • the Sampling Frequency Distribution Unit 3 is the interface between the transfer protocols 4 and the Estimation Unit 2. It supplies the terminal's own sampling frequency to the appropriate protocols, which for example can be TCP/IP protocols, and receives the sampling frequency from other terminals (not illustrated) for further distribution to the receiver 5.

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  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Multimedia (AREA)
  • Computer Hardware Design (AREA)
  • Synchronisation In Digital Transmission Systems (AREA)
  • Communication Control (AREA)
  • Data Exchanges In Wide-Area Networks (AREA)
  • Use Of Switch Circuits For Exchanges And Methods Of Control Of Multiplex Exchanges (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)

Abstract

L'invention porte sur un procédé et un appareil servant au transfert d'informations entre au moins deux émetteurs/récepteurs dans un système de télécommunications. Les données des informations sont transmises par l'émetteur d'un émetteur/récepteur au récepteur d'un ou plusieurs émetteurs/récepteurs sous forme de signaux numériques présentant une fréquence d'échantillonnage donnée qui sont reproduits par le récepteur d'une manière contrôlée. L'invention se caractérise principalement par une estimation de la cadence d'échantillonnage de l'émetteur de la partie émettrice de l'émetteur/récepteur, puis par la transmission de ladite estimation à la partie réceptrice d'un autre émetteur/récepteur, puis par le contrôle de la reproduction des données d'information par la partie réceptrice en utilisant l'estimation de la cadence d'échantillonnage pour éviter les retards et les interruptions dans la reproduction. L'invention est spécialement adaptée aux réseaux à base de paquets où les informations sont transmises entre les émetteurs/récepteurs du système de télécommunications sous forme de trames de paquets de données telles que des trames audio.
PCT/SE2000/000898 1999-07-08 2000-05-05 Procede de transfert d'information dans un systeme de telecommunications WO2001005076A1 (fr)

Priority Applications (4)

Application Number Priority Date Filing Date Title
EP00928087A EP1198910B1 (fr) 1999-07-08 2000-05-05 Procede de transfert d'information dans un systeme de telecommunications
DE60039763T DE60039763D1 (de) 1999-07-08 2000-05-05 Verfahren zum senden von informationen in einem telekommunikationssystem
AU46374/00A AU4637400A (en) 1999-07-08 2000-05-05 Method for sending information in a telecommunication system
JP2001509191A JP4553537B2 (ja) 1999-07-08 2000-05-05 電気通信システムにおける情報送信方法

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
SE9902630-4 1999-07-08
SE9902630A SE521462C2 (sv) 1999-07-08 1999-07-08 Förfarande och anordning för sändning av information i ett telekommunikationssystem

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WO2001005076A1 true WO2001005076A1 (fr) 2001-01-18

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EP (1) EP1198910B1 (fr)
JP (1) JP4553537B2 (fr)
CN (1) CN1268080C (fr)
AR (1) AR028150A1 (fr)
AU (1) AU4637400A (fr)
DE (1) DE60039763D1 (fr)
SE (1) SE521462C2 (fr)
WO (1) WO2001005076A1 (fr)

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JP4553537B2 (ja) 2010-09-29
EP1198910B1 (fr) 2008-08-06
CN1372732A (zh) 2002-10-02
US6870876B1 (en) 2005-03-22
DE60039763D1 (de) 2008-09-18
SE521462C2 (sv) 2003-11-04
SE9902630L (sv) 2001-01-09
AR028150A1 (es) 2003-04-30
SE9902630D0 (sv) 1999-07-08
EP1198910A1 (fr) 2002-04-24
JP2003504945A (ja) 2003-02-04
CN1268080C (zh) 2006-08-02
AU4637400A (en) 2001-01-30

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